Hello mailinglist,<br><br>I want to connect Asterisk with OpenBTS and make a call with a mobile phone. <br><br>I use:<br>Ubuntu 11.10 + Kernel 3.0.22<br>GnuRadio 3.3.0<br>Asterisk 1.8.13<br>OpenBTS 2.8<br>Nokia Mobile Phone<br>
<br>OpenBTS works and I can send sms from the OpenBTS server to the <br>mobile phone. What I also need is a call between Asterisk and OpenBTS.<br><br>I have also two soft phones which works with Asterisk. And also OpenBSC<br>
is working with Asterisk successfully (OpenBSC is another project).<br><br>Perhaps you can help me because I think it is an issue with Asterisk.<br><br><br>sip.conf:<br><blockquote style="margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote">
<font size="1">;SIP-Phones (Twinkle)<br>[user1]<br>callerid = 6000<br>username = 6000<br>secret = 6000<br>canreinvite = no<br>type = friend<br>context = phones<br>allow = all<br>host = dynamic<br>dtmfmode = info<br><br>[user2]<br>
callerid = 6001<br>username = 6001<br>secret = 6001<br>canreinvite = no<br>type = friend<br>context = phones<br>allow = all<br>host = dynamic<br>dtmfmode = info<br><br>; Mobile phone<br>[123456789101112]<br>callerid = 6201<br>
username = 6201<br>secret = 6201<br>canreinvite = no<br>type = friend<br>context = sip_external<br>;context = open-bts<br>disallow = all<br>allow = gsm<br>host = 192.168.0.102<br>domain = 192.168.0.102<br>dtmfmode = info<br>
</font></blockquote><br>extensions.conf<br><blockquote style="margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote"><font size="1">[internal]<br>exten => s,1,Verbose(1|Echo test application)<br>
exten => s,n,Echo()<br>exten => s,n,Hangup()<br>exten => 6000,1,Verbose(1|Extension 6000)<br>exten => 6000,n,Dial(SIP/user1,30)<br>exten => 6000,n,Hangup()<br>exten => 6001,1,Verbose(1|Extension 6001)<br>
exten => 6001,n,Dial(SIP/user2,30)<br>exten => 6001,n,Hangup()<br><br>[phones]<br>include => internal<br>include => default<br><br>[open-bts]<br>exten => 6002,1,Playback(demo-echotest)<br>exten => 6002,n,Echo<br>
exten => 6002,n,Playback(demo-echodone)<br>exten => 6002,n,HangUp<br><br>[sip_external]<br>
exten => 6201,1,Macro(dialGSM,123456789101112)<br><br>[macro-dialGSM]<br>exten => s,1,Dial(SIP/${ARG1},20)<br>exten => s,n,Goto(s-${DIALSTATUS},1)<br>exten => s-CANCEL,1,Hangup<br>exten => s-NOANSWER,1,Hangup<br>
exten => s-BUSY,1,Busy(30)<br>exten => s-CONGESTION,1,Congestion (30)<br>exten => s-CHANUNAVAIL,1,Read(extension_digits,pbx-invalid)<br>exten => s-CHANUNAVAIL,n,GoTo(open-bts,${extension_digits},1)<br></font></blockquote>
I have tried both contexts, [open-bts] and [sip_external] and both don't work<br><br><br>If I want to call the mobile phone (6201) with a Twinkle soft phone (6000) <br>I get following message in the CLI-window from Asterisk:<br>
<blockquote style="margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote"><font size="1"> == Using SIP RTP CoS mark 5<br> -- Executing [6201@DLPN_DialPlan1:1] Macro("SIP/6000-00000013", "stdexten,6201,SIP/6201") in new stack<br>
-- Executing [s@macro-stdexten:1] Set("SIP/6000-00000013", "__DYNAMIC_FEATURES=") in new stack<br><b> [Jul 12 12:14:29] WARNING[7092]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input:<br>
= 1<br> ^<br> [Jul 12 12:14:29] WARNING[7092]: ast_expr2.fl:472 ast_yyerror: If you have questions, please refer to <a href="https://wiki.asterisk.org/wiki/display/AST/Channel+Variables" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Channel+Variables</a><br>
-- Executing [s@macro-stdexten:2] GotoIf("SIP/6000-00000013", "?5:3") in new stack<br> -- Goto (macro-stdexten,s,3)<br> -- Executing [s@macro-stdexten:3] Dial("SIP/6000-00000013", "SIP/6201,20,") in new stack<br>
[Jul 12 12:14:29] WARNING[7092]: app_dial.c:2274 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)<br> == Everyone is busy/congested at this time (1:0/0/1)</b><br> -- Executing [s@macro-stdexten:4] Goto("SIP/6000-00000013", "s-CHANUNAVAIL,1") in new stack<br>
-- Goto (macro-stdexten,s-CHANUNAVAIL,1)<br> -- Executing [s-CHANUNAVAIL@macro-stdexten:1] Goto("SIP/6000-00000013", "s-NOANSWER,1") in new stack<br> -- Goto (macro-stdexten,s-NOANSWER,1)<br>
-- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("SIP/6000-00000013", "6201,u") in new stack<br> -- <SIP/6000-00000013> Playing 'vm-theperson.gsm' (language 'en')<br>
-- <SIP/6000-00000013> Playing 'digits/6.gsm' (language 'en')<br> -- <SIP/6000-00000013> Playing 'digits/2.gsm' (language 'en')<br> -- <SIP/6000-00000013> Playing 'digits/0.gsm' (language 'en')<br>
-- <SIP/6000-00000013> Playing 'digits/1.gsm' (language 'en')<br> -- <SIP/6000-00000013> Playing 'vm-isunavail.gsm' (language 'en')<br> -- <SIP/6000-00000013> Playing 'vm-intro.gsm' (language 'en')<br>
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'SIP/6000-00000013' in macro 'stdexten'<br> == Spawn extension (DLPN_DialPlan1, 6201, 1) exited non-zero on 'SIP/6000-00000013'<br>
</font></blockquote><br><br>*CLI> sip show peers<br><blockquote style="margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote"><font size="1"> Name/username Host Dyn Forcerport ACL Port Status <br>
<b> 123456789101112/6201 192.168.0.102 N 5060 Unmonitored</b><br> 6000/6000 192.168.0.102 D N 5061 Unmonitored<br>
6001/6001 192.168.0.102 D N 5061 Unmonitored<br> (...)<br> user1/6000 (Unspecified) D N 0 Unmonitored<br>
user2/6001 (Unspecified) D N 0 Unmonitored <br></font></blockquote><br>*CLI> sip show peer 123456789101112<br><blockquote style="margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote">
<font size="1"> * Name : 123456789101112<br> Secret : <Set><br> MD5Secret : <Not set><br> Remote Secret: <Not set><br><b> Context : sip_external</b><br> Subscr.Cont. : device-hints<br>
Language :<br> AMA flags : Unknown<br> Transfer mode: open<br> CallingPres : Presentation Allowed, Not Screened<br> Callgroup :<br> Pickupgroup :<br> MOH Suggest :<br> Mailbox :<br>
<b> VM Extension : asterisk</b><br> LastMsgsSent : 32767/65535<br> Call limit : 0<br> Max forwards : 0<br> Dynamic : No<br><b> Callerid : "" <6201></b><br> MaxCallBR : 384 kbps<br>
Expire : -1<br> Insecure : no<br> Force rport : Yes<br> ACL : No<br> DirectMedACL : No<br> T.38 support : No<br> T.38 EC mode : Unknown<br> T.38 MaxDtgrm: -1<br>
DirectMedia : No<br> PromiscRedir : No<br> User=Phone : No<br> Video Support: No<br> Text Support : No<br> Ign SDP ver : No<br> Trust RPID : No<br> Send RPID : No<br> Subscriptions: Yes<br>
Overlap dial : No<br> DTMFmode : info<br> Timer T1 : 500<br> Timer B : 32000<br> <b>ToHost : 192.168.0.102<br> Addr->IP : <a href="http://192.168.0.102:5060" target="_blank">192.168.0.102:5060</a></b><br>
Defaddr->IP : (null)<br> Prim.Transp. : UDP<br> Allowed.Trsp : UDP<br> Def. Username: 6201<br> SIP Options : (none)<br> Codecs : 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719)<br>
Codec Order : (none)<br> Auto-Framing : No<br> Status : Unmonitored<br> Useragent :<br> Reg. Contact :<br> Qualify Freq : 60000 ms<br> Sess-Timers : Accept<br> Sess-Refresh : uas<br>
Sess-Expires : 1800 secs<br> Min-Sess : 90 secs<br><b> RTP Engine : asterisk</b><br> Parkinglot :<br> Use Reason : No<br> Encryption : No<br></font></blockquote><br>Asterisk log file (path: /var/log/asterisk/cdr-csv/Master.csv):<br>
<blockquote style="margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote"><font size="1"> "","6000","6201","DLPN_DialPlan1","""6000"" <6000>","SIP/6000-00000013","","VoiceMail","6201,u","2012-07-12 10:14:29","2012-07-12 10:14:29","2012-07-12 10:14:35",6,6,"ANSWERED","DOCUMENTATION","1342088069.31",""<br>
</font></blockquote><br><br><br>If you need more informations write me and I will give you. It would be very <br>appreciated if some of you can help me or has an idea how I can fix this erorr. <br><br>Best regards and thanks for helping.<br>
Ellen<br>